1. Technical Field
The invention relates to a method and apparatus for encoding digital signals for transmission and/or storage. The invention is especially, but not exclusively, applicable to the encoding of digital signals for transmission via communications channels, such as twisted wire pair subscriber loops in telecommunications systems, or to storage of signals in or on a storage medium, such as video signal recordings, audio recordings, data storage in computer systems, and so on.
2. Background Art
Embodiments of the invention are especially applicable to Asynchronous Transfer Mode (ATM) telecommunications systems. Such systems are now available to transmit millions of data bits in a single second and are expected to turn futuristic interactive concepts into exciting realities within the next few years. However, deployment of ATM is hindered by expensive port cost and the cost of running an optical fiber from an ATM switch to the customer-premises using an architecture known as Fiber-to-the-home. Running ATM traffic in part of the subscriber loop over existing copper wires would reduce the cost considerably and render the connection of ATM to customer-premises feasible.
The introduction of ATM signals in the existing twisted-pair subscriber loops leads to a requirement for bit rates which are higher than can be achieved with conventional systems in which there is a tendency, when transmitting at high bit rates, to lose a portion of the signal, typically the higher frequency part, causing the signal quality to suffer significantly. This is particularly acute in two-wire subscriber loops, such as socalled twisted wire pairs. Using quadrature amplitude modulation (QAM), it is possible to meet the requirements for Asymmetric Digital Subscriber Loops (ADSL), involving rates as high as 1.5 megabits per second for loops up to 3 kilometers long with specified error rates. It is envisaged that ADSL systems will allow rates up to about 8 megabits per second over 1 kilometer loops. Nevertheless, these rates are still considered to be too low, given that standards currently proposed for ATM basic subscriber access involve rates of about 26 megabits per second.
QAM systems tend to operate at the higher frequency bands of the channel, which is particularly undesirable for two-wire subscriber loops where attenuation and cross-talk are worse at the higher frequencies. It has been proposed, therefore, to use frequency division modulation (FDM) to divide the transmission system into a set of frequency-indexed sub-channels. The input data is partitioned into temporal blocks, each of which is independently modulated and transmitted in a respective one of the sub-channels. One such system, known as discrete multi-tone transmission (DMT), is disclosed in U.S. Pat. No. 5,479,447 issued December 1995 and in an article entitled xe2x80x9cPerformance Evaluation of a Fast Computation Algorithm for the DMT in High-Speed Subscriber Loopxe2x80x9d, IEEE Journal on Selected Areas in Communications, Vol. 13, No. 9, December 1995 by I. Lee et al. Specifically, U.S. Pat. No. 5,479,447 discloses a method and apparatus for adaptive, variable bandwidth, high-speed data transmission of a multi-carrier signal over a digital subscriber loop. The data to be transmitted is divided into multiple data streams which are used to modulate multiple carriers. These modulated carriers are converted to a single high speed signal by means of IFFT (Inverse Fast Fourier Transform) before transmission. At the receiver, Fast Fourier Transform (FFT) is used to split the received signal into modulated carriers which are demodulated to obtain the original multiple data streams.
Such a DMT system is not entirely satisfactory for use in two-wire subscriber loops which are very susceptible to noise and other sources of degradation which could result in one or more sub-channels being lost. If only one sub-channel fails, perhaps because of transmission path noise, the total signal is corrupted and either lost or, if error detection is employed, may be retransmitted. It has been proposed to remedy this problem by adaptively eliminating noisy sub-channels, but to do so would involve very complex circuitry.
A further problem with DMT systems is poor separation between sub-channels. In U.S. Pat. No. 5,497,398 issued March 1996, M. A. Tzannes and M. C. Tzannes proposed ameliorating the problem of degradation due to sub-channel loss, and obtaining superior burst noise immunity, by replacing the Fast Fourier Transform with a lapped transform, thereby increasing the difference between the main lobe and side lobes of the filter response in each sub-channel. The lapped transform may comprise wavelets, as disclosed by M. A. Tzannes, M. C. Tzannes and H. L. Resnikoff in an article xe2x80x9cThe DWMT: A Multicarrier Transceiver for ADSL using M-band Waveletsxe2x80x9d, ANSI Standard Committee T1E1.4 Contribution 93-067, March 1993 and by S. D. Sandberg, M. A. Tzannes in an article xe2x80x9cOverlapped Discrete Multitone Modulation for High Speed Copper Wire Communicationsxe2x80x9d, IEEE Journal on Selected Areas in Comm., Vol. 13, No. 9, pp. 1571-1585, Dec. 1995, such systems being referred to as xe2x80x9cDiscrete Wavelet Multitone (DWNIT).
A disadvantage of both DMT and DWMT systems is that they typically use a large number of sub-channels, for example 256 or 512, which leads to complex, costly equipment and equalization and synchronization difficulties. These difficulties are exacerbated if, to take advantage of the better characteristics of the two-wire subscriber loop at lower frequencies, the number of bits transmitted at the lower frequencies is increased and the number of bits transmitted at the higher frequencies reduced correspondingly.
It is known to use sub-band filtering to process digital audio signals prior to recording on a storage medium, such as a compact disc. Thus, U.S. Pat. No. No. 5,214,678 (Rault et al) discloses an arrangement for encoding audio signals and the like into a set of sub-band signals using a commutator and a plurality of analysis filters, which could be combined. Rault et al use recording means which record the sub-band signals as multiple, distinct tracks. This is not entirely satisfactory because each sub-band signal would require its own recording head or, if applied to transmission, its own transmission channel.
It is also known to use sub-band filtering for compression of audio signals, as disclosed by C. Heegard and T. Shamoon in xe2x80x9cHigh-Fidelity Audio Compression: Fractional-Band Waveletxe2x80x9d, 1992 IEEE Conference on Acoustics, Speech and Signal Processing, 23-26 March 1992, New York.
In an article entitled xe2x80x9cWavelet-Coded Image Transmission Over Land Mobile Radio Channels, IEEE Global Telecommunications Conference, 6-9 December 1992, New York, You-Jong Liu et al disclosed the use of two-dimensional wavelet decomposition to convert an image into sub-images. The sub-images were quantized to produce digital numeric representations which were transmitted.
U.S. Pat. No. 5,161,210 (Druyvesteyn) discloses a similar analysis technique to that disclosed by Rault et al but, in this case, the sub-band signals are combined by means of a synthesis filter before recordal. The input audio signal first is analyzed, and an identification signal is mixed with each of the sub-band signals. The sub-band signals then are recombined using a synthesis filter. The technique ensures that the identification signal cannot be removed simply by normal filtering. The frequency spectrum of the recombined signal is substantially the same as that of the input signal, so it would still be susceptible to corruption by loss of the higher frequency components. The corresponding decoder also comprises an analysis filter and a synthesis filter. Consequently, the apparatus is very complex and would involve delays which would be detrimental in high speed transmission systems.
It is desirable to combine the sub-band signals in such a way as to reduce the risk of corruption resulting from part of the signal being lost or corrupted during transmission and/or storage.
It should be noted that, although Rault et al use the term xe2x80x9canalysis filterxe2x80x9d in their specification, in this specification the term xe2x80x9canalysis filterxe2x80x9d will be used to denote a device which decomposes a signal into a plurality of sub-band signals in such a way that the original signal can be reconstructed using a complementary synthesis filter.
The present invention seeks to eliminate, or at least mitigate, the disadvantages of these known systems and has for its object to provide an improved method and apparatus for encoding signals for transmission and/or storage.
According to one aspect of the invention, apparatus for encoding an input signal for transmission or storage and decoding such encoded signal to reconstruct the input signal, comprising an encoder (11) for encoding a digital input signal for transmission or storage and a decoder (13) for decoding such encoded signal to reconstruct the input signal, the encoder comprising analysis filter bank means (21;51) for analyzing the input signal into a plurality of sub-band signals, each sub-band centered at a respective one of a corresponding plurality of frequencies and the decoder comprising synthesis filter bank means (33;67) complementary to said analysis filter bank means for producing a decoded signal corresponding to the input signal, characterized in that: the encoder (11) comprises
(i) interpolation means (52) for interpolating of the plurality of sub-band signals to provide a plurality of interpolated signals each occupying the same frequency band as the others; and
(ii) combining means (23;58) for combining the interpolated sub-band signals to form
the encoded signal for transmission or storage; and the decoder (13) comprises
(iii) means (31;610, 611, 612) for extracting the interpolated sub-band signals from the received or recorded encoded signal;
(iv) decimator means (66) for decimating each of the plurality of extracted interpolated sub-band signals to remove the interpolated values and applying the decimated signals to the synthesis filter bank means, the synthesis filter bank means processing the plurality of decimated sub-band signals to reconstruct said input signal.
According to second and third aspects of the invention, there are provided the encoder per se and the decoder per se of the apparatus.
The analysis filter means may be uniform, for example an M-band filter bank or Short-time Fast Fourier Transform unit; or non-uniform, for example a xe2x80x9cmultiresolutionxe2x80x9d filter bank such as an octave-band or dyadic filter bank implementing discrete wavelet transform (DWT) which will produce sub-bands having different bandwidths, typically each half the width of its neighbour.
The interpolation rate may be such that the resulting interpolated sub-band signals all have the same rate.
The interpolation rate will be chosen according to the requirements of a particular transmission channel or storage means but typically will be of the order of 1:8 or more.
The interpolation means may comprise an upsampler, for interpolating intervals between actual values, and filter means, for example Raise Cosine filter means, for determining values between the actual samples and inserting them at the appropriate intervals.
Usually, when used with digital signals, sub-band analysis filter banks create sub-band signals which occupy a wide spectrum as compared with the original signal, which makes modulation difficult. Interpolating and smoothing the sub-band signals advantageously band-limits the spectrum of the sub-band signals, permitting modulation by a variety of techniques, for example Double or Single Sideband Amplitude Modulation, Quadrature Amplitude Modulation (QAM), Carrier Amplitude/Phase modulation (CAP), and so on.
According to a fourth aspect of the invention, there is provided a method of encoding an input signal for transmission or storage and decoding such encoded signal to reconstruct the input signal, the encoding of the input signal comprising the steps of using analysis filter bank means to analyze the input signal into a plurality of sub-band signals, each sub-band centered at a respective one of a corresponding plurality of frequencies, characterized in that the encoding comprises the steps of:
(i) interpolating each of the plurality of sub-band signals to provide a corresponding plurality of interpolated sub-band signals each occupying the same frequency band as the others; and
(ii) combining the interpolated sub-band signals to form the encoded signal for transmission or storage;
and the decoding of the encoded signal comprises the steps of:
(iii) extracting the plurality of interpolated sub-band signals from the received or recorded encoded signal;
(iv) decimating each of the plurality of extracted interpolated sub-band signals to remove values interpolated during encoding; and
(v) using synthesis filter bank means complementary to said analysis filter bank means, processing the plurality of decimated sub-band signals to produce a decoded signal corresponding to the input signal.
According to fifth and sixth aspects of the invention, there are provided the method of encoding per se and the method of decoding per se.
In embodiments of any of the above aspects of the invention which use Discrete Wavelet Transform, the digital input signal may be divided into segments and the discrete wavelet transform used to transform successive segments of the digital signal.
The foregoing and other objects, features, aspects and advantages of the present invention will become more apparent from the following detailed description of preferred embodiments of the invention which are described by way of example only with reference to the accompanying drawings.